Mercurial > libavcodec.hg
view ra144.c @ 6872:60236c4f0b76 libavcodec
cosmetics: indent
author | jbr |
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date | Sun, 25 May 2008 16:20:06 +0000 |
parents | 62ce6173d0ea |
children | c3a92478875d |
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/* * Real Audio 1.0 (14.4K) * Copyright (c) 2003 the ffmpeg project * * This file is part of FFmpeg. * * FFmpeg is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * FFmpeg is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "avcodec.h" #include "bitstream.h" #include "ra144.h" #define NBLOCKS 4 /* number of segments within a block */ #define BLOCKSIZE 40 /* (quarter) block size in 16-bit words (80 bytes) */ #define HALFBLOCK 20 /* BLOCKSIZE/2 */ #define BUFFERSIZE 146 /* for do_output */ /* internal globals */ typedef struct { unsigned int oldval; /* the swapped buffers */ unsigned int swapbuffers[4][10]; unsigned int *swapbuf1; unsigned int *swapbuf2; unsigned int *swapbuf1alt; unsigned int *swapbuf2alt; unsigned int buffer[5]; unsigned short int buffer_2[148]; } Real144_internal; static int ra144_decode_init(AVCodecContext * avctx) { Real144_internal *glob = avctx->priv_data; glob->swapbuf1 = glob->swapbuffers[0]; glob->swapbuf2 = glob->swapbuffers[1]; glob->swapbuf1alt = glob->swapbuffers[2]; glob->swapbuf2alt = glob->swapbuffers[3]; return 0; } /* lookup square roots in table */ static int t_sqrt(unsigned int x) { int s = 0; while (x > 0xfff) { s++; x = x >> 2; } return (ff_sqrt(x << 20) << s) << 2; } /* do 'voice' */ static void do_voice(const int *a1, int *a2) { int buffer[10]; int *b1 = buffer; int *b2 = a2; int x, y; for (x=0; x < 10; x++) { b1[x] = a1[x] << 4; for (y=0; y < x; y++) b1[y] = ((a1[x] * b2[x-y-1]) >> 12) + b2[y]; FFSWAP(int *, b1, b2); } for (x=0; x < 10; x++) a2[x] >>= 4; } /* rotate block */ static void rotate_block(const short *source, short *target, int offset) { int i=0, k=0; source += BUFFERSIZE - offset; while (i<BLOCKSIZE) { target[i++] = source[k++]; if (k == offset) k = 0; } } /* inverse root mean square */ static int irms(const short *data, int factor) { unsigned int i, sum = 0; for (i=0; i < BLOCKSIZE; i++) sum += data[i] * data[i]; if (sum == 0) return 0; /* OOPS - division by zero */ return (0x20000000 / (t_sqrt(sum) >> 8)) * factor; } /* multiply/add wavetable */ static void add_wav(int n, int f, int m1, int m2, int m3, const short *s1, const int8_t *s2, const int8_t *s3, short *dest) { int a = 0; int b, c, i; const int16_t *ptr; const uint8_t *ptr2; ptr = wavtable1[n]; ptr2 = wavtable2[n]; if (f) a = (ptr[0] * m1) >> (ptr2[0] + 1); b = (ptr[1] * m2) >> (ptr2[1] + 1); c = (ptr[2] * m3) >> (ptr2[2] + 1); for (i=0; i < BLOCKSIZE; i++) dest[i] = ((*(s1++)) * a + (*(s2++)) * b + (*(s3++)) * c) >> 12; } static void final(const short *i1, const short *i2, void *out, int *statbuf, int len) { int x, i; unsigned short int work[50]; short *ptr = work; memcpy(work, statbuf,20); memcpy(work + 10, i2, len * 2); for (i=0; i<len; i++) { int sum = 0; int new_val; for(x=0; x<10; x++) sum += i1[9-x] * ptr[x]; sum >>= 12; new_val = ptr[10] - sum; if (new_val < -32768 || new_val > 32767) { memset(out, 0, len * 2); memset(statbuf, 0, 20); return; } ptr[10] = new_val; ptr++; } memcpy(out, work+10, len * 2); memcpy(statbuf, work + 40, 20); } static unsigned int rms(const int *data, int f) { int x; unsigned int res = 0x10000; int b = 0; for (x=0; x<10; x++) { res = (((0x1000000 - (*data) * (*data)) >> 12) * res) >> 12; if (res == 0) return 0; if (res > 0x10000) return 0; /* We're screwed, might as well go out with a bang. :P */ while (res <= 0x3fff) { b++; res <<= 2; } data++; } if (res > 0) res = t_sqrt(res); res >>= (b + 10); res = (res * f) >> 10; return res; } /* do quarter-block output */ static void do_output_subblock(Real144_internal *glob, const unsigned short *gsp, unsigned int gval, signed short *output_buffer, GetBitContext *gb) { unsigned short int buffer_a[40]; unsigned short int *block; int e, f, g; int a = get_bits(gb, 7); int d = get_bits(gb, 8); int b = get_bits(gb, 7); int c = get_bits(gb, 7); if (a) { a += HALFBLOCK - 1; rotate_block(glob->buffer_2, buffer_a, a); } e = ((ftable1[b] >> 4) * gval) >> 8; f = ((ftable2[c] >> 4) * gval) >> 8; if (a) g = irms(buffer_a, gval) >> 12; else g = 0; memmove(glob->buffer_2, glob->buffer_2 + BLOCKSIZE, (BUFFERSIZE - BLOCKSIZE) * 2); block = glob->buffer_2 + BUFFERSIZE - BLOCKSIZE; add_wav(d, a, g, e, f, buffer_a, etable1[b], etable2[c], block); final(gsp, block, output_buffer, glob->buffer, BLOCKSIZE); } static int dec1(signed short *decsp, const int *data, const int *inp, int f) { short *ptr,*end; end = (ptr = decsp) + 30; while (ptr < end) *(ptr++) = *(inp++); return rms(data, f); } static int eq(const short *in, int *target) { int retval = 0; int b, c, i; unsigned int u; int buffer1[10]; int buffer2[10]; int *bp1 = buffer1; int *bp2 = buffer2; for (i=0; i < 10; i++) buffer2[i] = in[i]; u = target[9] = bp2[9]; if (u + 0x1000 > 0x1fff) return 0; /* We're screwed, might as well go out with a bang. :P */ for (c=8; c >= 0; c--) { if (u == 0x1000) u++; if (u == 0xfffff000) u--; b = 0x1000-((u * u) >> 12); if (b == 0) b++; for (u=0; u<=c; u++) bp1[u] = ((bp2[u] - ((target[c+1] * bp2[c-u]) >> 12)) * (0x1000000 / b)) >> 12; target[c] = u = bp1[c]; if ((u + 0x1000) > 0x1fff) retval = 1; FFSWAP(int *, bp1, bp2); } return retval; } static int dec2(signed short *decsp, const int *data, const int *inp, int f, const int *inp2, int l) { unsigned const int *ptr1,*ptr2; int work[10]; int a,b; int x; int result; unsigned short *sptr = decsp; if(l + 1 < NBLOCKS / 2) a = NBLOCKS - (l + 1); else a = l + 1; b = NBLOCKS - a; ptr1 = inp; ptr2 = inp2; for (x=0; x<30; x++) *(sptr++) = (a * (*ptr1++) + b * (*ptr2++)) >> 2; result = eq(decsp, work); if (result == 1) { return dec1(decsp, data, inp, f); } else { return rms(work, f); } } /* Uncompress one block (20 bytes -> 160*2 bytes) */ static int ra144_decode_frame(AVCodecContext * avctx, void *vdata, int *data_size, const uint8_t * buf, int buf_size) { static const uint8_t sizes[10] = {6, 5, 5, 4, 4, 3, 3, 3, 3, 2}; unsigned int gbuf1[4]; unsigned short gbuf2[4][30]; unsigned int a, c; int i; int16_t *data = vdata; unsigned int val; Real144_internal *glob = avctx->priv_data; GetBitContext gb; if(buf_size < 20) { av_log(avctx, AV_LOG_ERROR, "Frame too small (%d bytes). Truncated file?\n", buf_size); return buf_size; } init_get_bits(&gb, buf, 20 * 8); for (i=0; i<10; i++) // "<< 1"? Doesn't this make one value out of two of the table useless? glob->swapbuf1[i] = decodetable[i][get_bits(&gb, sizes[i]) << 1]; do_voice(glob->swapbuf1, glob->swapbuf2); val = decodeval[get_bits(&gb, 5) << 1]; // Useless table entries? a = t_sqrt(val*glob->oldval) >> 12; gbuf1[0] = dec2(gbuf2[0], glob->swapbuf1alt, glob->swapbuf2alt, glob->oldval, glob->swapbuf2, 0); if (glob->oldval < val) { gbuf1[1] = dec2(gbuf2[1], glob->swapbuf1, glob->swapbuf2, a, glob->swapbuf2alt, 1); } else { gbuf1[1] = dec2(gbuf2[1], glob->swapbuf1alt, glob->swapbuf2alt, a, glob->swapbuf2, 1); } gbuf1[2] = dec2(gbuf2[2], glob->swapbuf1, glob->swapbuf2, val, glob->swapbuf2alt, 2); gbuf1[3] = dec1(gbuf2[3], glob->swapbuf1, glob->swapbuf2, val); /* do output */ for (c=0; c<4; c++) { do_output_subblock(glob, gbuf2[c], gbuf1[c], data, &gb); for (i=0; i<BLOCKSIZE; i++) { *data = av_clip_int16(*data << 2); data++; } } glob->oldval = val; FFSWAP(unsigned int *, glob->swapbuf1alt, glob->swapbuf1); FFSWAP(unsigned int *, glob->swapbuf2alt, glob->swapbuf2); *data_size = 2*160; return 20; } AVCodec ra_144_decoder = { "real_144", CODEC_TYPE_AUDIO, CODEC_ID_RA_144, sizeof(Real144_internal), ra144_decode_init, NULL, NULL, ra144_decode_frame, .long_name = "RealAudio 1.0 (14.4K)", };